The present invention relates generally to providing quality of service assurance over non-wireless portions of a wireless voice over Internet Protocols (VOIP) system, and to locating and connecting to destination devices outside of a serving cell site.
Wireless telephones, such as but not limited to wireless telephones that communicate using Code Division Multiple Access (CDMA) spread spectrum modulation techniques, communicate over the air with system infrastructure using wireless telephone over-the-air communication protocols, e.g., the CDMA protocols known as IS-95A, IS-95B, and IS-2000. The system infrastructure, which can include base stations (BTS), base station controllers (BSC), and other components, connects the wireless telephone to another communication device, such as a through land line or another wireless communication system.
With the growth of the Internet, computer-to-computer communication using Internet Protocols (IP) has become ubiquitous. Furthermore, it has become desirable not only to facilitate computer data communication using IP, but to facilitate voice communication using IP as well. As but one advantage afforded by using IP in a telephony infrastructure, much hardware such as switches can be eliminated, and existing computers and software can be used instead, reducing cost. To this end, so-called voice over IP (VOIP) has been introduced. As recognized herein, however, VOIP data is latency-sensitive (i.e., it is desirable that voice data not be unduly delayed in transmission between people conversing with each other).
To connect an originating device with a destination device, an auxiliary IP-based protocol known as Session Initiation Protocol (SIP) is used. Using a protocol such as SIP, a wireless device is associated with an IP address and a unique identifying alphanumeric packet address, such as xe2x80x9csip://MY_PHONE@qualcomm.com.xe2x80x9d An SIP server functions as a directory of endpoints and their associated IP addresses and packet addresses. Accordingly, to participate in IP-based communication such as VOIP, a device must register its IP address with the SIP server. When an originating device requests a connection to an IP-based destination device, the SIP server either gives the destination IP address to the originating device, or it establishes a connection with the destination device and then acts as a proxy for the originating and destination devices.
In any case, when an originating device places a call to a destination device in the same service system (an xe2x80x9cintrasystemxe2x80x9d call), an SIP server that may be associated with the system knows both addresses and consequently establishes, perhaps using one or more options such as an xe2x80x9cencryptionxe2x80x9d option, an IP connection between the two devices. On the other hand, if the destination device is not in the same wireless service system as the originating device (an xe2x80x9cintersystemxe2x80x9d call), its address will not appear in the SIP server database, and the SIP server consequently forwards the call request to other SIP servers until the destination address is located or until the request times out.
As recognized by the present invention, because the transmission of packets is IP-based, intrasystem calls and intersystem calls in particular can result in transmitting packets over publicly-accessible portions of the Internet, resulting in unpredictable and potentially fluctuating delays. In the case of latency-intensive applications such as VOIP, this can severely compromise performance, since a calling party might experience undue delays in having his voice heard by a called party and vice-versa.
Standard voice-over-IP telephony calls originated to destinations within the same service system are connected to the destination party using Internet Protocol (IP) routing techniques with the assistance of a SIP server, as noted above. VOIP telephone calls originated to destinations outside of the originating service system to destinations on the PSTN require format conversion at the boundary of the originating packet system (the xe2x80x9cintranetxe2x80x9d) and the PSTN. A xe2x80x9cVOIP gatewayxe2x80x9d is a device well known in the art that converts between VOIP and PSTN formats. In order to perform this conversion, the VOIP gateway requires knowledge of the voice encoding and voice call signaling used within the originating IP system (intranet). Hence, VOIP systems employ standardized voice encoding techniques. Another case of VOIP calls of increasing interest are encrypted VOIP calls. Encrypted VOIP calls use voice encoding involving exchange of encryption establishment information between the two (or more) participating phones, and subsequent exchange of encrypted voice packets. These encrypted voice packets are not understood by VOIP gateways, and must be exchanged between the participating phones using a data network. In other words, an end-to-end data connection is required between the participating phones. Having made the above critical observations, the present invention provides the solutions disclosed herein.
A voice over Internet (VOIP) system includes an IP-based infrastructure component communicating with a wireless communication originating device. An SIP server communicates with the infrastructure component, and a modem bank is associated with the SIP server and is connected to the PSTN. The SIP server selectively uses the modem bank to instantiate a circuit-switched call from the originating device to a destination device. In a preferred embodiment, the SIP server instantiates circuit-switched calls only for intersystem calls requiring voice-call-latency characteristics and end-to-end data connectivity, as might be indicated by, e.g., an address or portion thereof of the destination device not being registered with the SIP server, and/or specific SIP call setup parameters. Thus, the SIP server does not instantiate a circuit-switched call for an intrasystem call, e.g., a call to a destination device currently served by the same intranet as the originating device.
In a particularly preferred non-limiting embodiment, the destination device has an IP address of the form xe2x80x9csip://DN@service.comxe2x80x9d, and the circuit switched call is instantiated using the DN portion of the IP address. In one embodiment, the SIP server maps the DN to the modem bank and completes the call through the PSTN. Additional information conveyed, potentially including optional SIP call parameters, can be used to identify a VOIP call as a secure call requiring instantiation of the circuit switched call.
In another aspect, a method for VOIP includes receiving, from an originating wireless device in a first service system, a call request for a destination device that has an IP address. The method then determines whether the destination device is in the first service system. If the destination device is in the first service system, communication is established between the originating device and destination device within the first service system without using the public Internet. Otherwise, a circuit-switched call is established between the destination device and originating device using the PSTN, particularly if the call is encrypted. In neither case are Internet connections which potentially are subject to unpredictable delays, such as the publicly accessible portions of the Internet, used to complete the VOIP call.
In still another aspect, a computer program device includes means for receiving a call request at an SIP server. The call request is generated by a wireless communication originating device, and it indicates a destination device. The program device also includes means for determining whether the destination device is registered with the SIP server. Means are provided for transmitting IP packets between the destination device and originating device without using the public Internet when the destination device is registered with the SIP server. Also, however, means are provided for transmitting IP packets between the destination device and originating device using an instantiated circuit-switched connection when the destination device is not registered with the SIP server, and perhaps only when the call is encrypted.
The details of the present invention, both as to its structure and operation, can best be understood in reference to the accompanying drawings, in which like reference numerals refer to like parts, and in which: